[asterisk-dev] [Code Review]: Testsuite: Disallow MoH upon hold option

Mark Michelson reviewboard at asterisk.org
Mon Feb 25 14:52:05 CST 2013



> On Feb. 21, 2013, 4:59 p.m., opticron wrote:
> > /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_media_restrict.xml, lines 52-64
> > <https://reviewboard.asterisk.org/r/2337/diff/1/?file=33505#file33505line52>
> >
> >     This must include the same number of streams as were offered in the initial INVITE to be correct.

To expand a bit on this, a 200 OK for an INVITE should ALWAYS have an SDP, at least when the purpose of the INVITE is to set up a media session. Depending on the offer-answer model being used, it's possible for the INVITE or ACK not to have an SDP, but the 200 OK will always have one.


- Mark


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2337/#review7910
-----------------------------------------------------------


On Feb. 18, 2013, 2:30 p.m., Kevin Harwell wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2337/
> -----------------------------------------------------------
> 
> (Updated Feb. 18, 2013, 2:30 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> Tests to make sure the music on hold event is not triggered if the "discard_remote_hold_retrieval" option is set to "yes".  In the test multiple scenarios are ran where one SIP phone puts another SIP phone on hold by sending a re-INVITE with a modified SDP containing either a restricted audio direction, an IP address of 0.0.0.0, or a combination thereof.  This is tested both for a local RTP bridge, and a non-bridged scenario.
> 
> 
> This addresses bug ABE-2899.
>     https://issues.asterisk.org/jira/browse/ABE-2899
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_A.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_IP_media_restrict.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_IP_restrict.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_media_restrict.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/tests.yaml 3642 
> 
> Diff: https://reviewboard.asterisk.org/r/2337/diff
> 
> 
> Testing
> -------
> 
> Ran the test and made sure all scenarios passed.  Also set the discard_remote_hold_retrieval to "no" and made sure the test failed. 
> 
> 
> Thanks,
> 
> Kevin
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130225/bd418db1/attachment.htm>


More information about the asterisk-dev mailing list